How to Set Up Bitcall with Asterisk Using IP Authentication

A complete guide to connecting your Asterisk server to Bitcall using IP-to-IP Authentication. This is the professional standard for servers with a fixed public IP it’s faster, more stable, and doesn’t require SIP passwords or registration lines.

What You Will Need

• A VPS with a Static Public IP

• An active Bitcall account panel.bitcall.io

• SSH access to your server

Phase 1 : Whitelist your Server IP [Browser Step]

Before configuring Asterisk, you must tell Bitcall to trust your server's IP address.

1. Log in to panel.bitcall.io.

2. Go to IP Authentication in the sidebar.

3. Click Add IP.

4. Enter your server's Public IP (e.g., YOUR-SERVER-IP).

5. Click Save.

Phase 2 : Choose Your Driver [Terminal Step]

Option A : chan_sip Configuration (sip.conf)

Use this for older Asterisk versions or simple setups.

Steps:

1. Clear old registration lines: Run this command to remove any existing registration from sip.conf to avoid conflicts:

   sed -i '/register =>/d' /etc/asterisk/sip.conf

2. Append the peer block: Run this command to add the Bitcall IP block to the end of sip.conf:

cat >> /etc/asterisk/sip.conf << 'EOF'

[bitcall-ip]

type=peer

host=gateway-ip-here

context=from-bitcall

insecure=invite

qualify=yes

disallow=all

allow=ulaw

allow=alaw

nat=yes

EOF

⚠️ Replace gateway-ip-here with the Bitcall gateway IP provided in your panel.

3. Reload Asterisk: asterisk -rx "reload"

4. Verify: asterisk -rx "sip show peers"

   • You should see bitcall-ip with status OK.

Option B : PJSIP Configuration (pjsip.conf)

Use this for Asterisk 16+ (the modern standard).

Steps:

1. Add PJSIP blocks: Run this command to append the IP-to-IP blocks to pjsip.conf:

cat >> /etc/asterisk/pjsip.conf << 'EOF'

; === BitCall IP-to-IP Configuration ===

[bitcall-endpoint-ip]

type=endpoint

transport=transport-udp

aors=bitcall-aor-ip

context=from-bitcall

disallow=all

allow=ulaw

allow=alaw

[bitcall-aor-ip]

type=aor

contact=sip:gateway-ip-here

[bitcall-identify-ip]

type=identify

endpoint=bitcall-endpoint-ip

match=gateway-ip-here

EOF

⚠️ Replace gateway-ip-here with the Bitcall gateway IP.

2. Restart Asterisk: asterisk -rx "core restart now"

3. Verify: asterisk -rx "pjsip show endpoints"

   • You should see bitcall-endpoint-ip in status Not in use (Available).

TIP

To switch between chan_sip and PJSIP for IP-to-IP: Use the same noload commands in modules.conf as described in Article 1. The underlying logic is the same.

Expected Result

Asterisk sends calls directly to Bitcall without needing a login. Bitcall recognizes your server by its IP and routes the call instantly.

Make Your First Test Call [Terminal Step]

Since there is no "Registration" (State: Registered) for IP-to-IP, you test by making a call directly from the SSH terminal:

# For chan_sip (Option A):

asterisk -rx "channel originate SIP/bitcall-ip/YOUR-MOBILE-NUMBER application Playback hello-world"

# For PJSIP (Option B):

asterisk -rx "channel originate PJSIP/YOUR-MOBILE-NUMBER@bitcall-endpoint-ip application Playback hello-world"

⚠️ Replace YOUR-MOBILE-NUMBER with your actual phone number.

Troubleshooting

• Call Rejected (403 Forbidden): Ensure your Server IP is correctly added to the IP Authentication section in the Bitcall panel.

• Peer Status: UNKNOWN: Check the host= or contact= IP address. Ensure your server firewall allows traffic on UDP Port 5060.

• One-way audio: Double-check NAT settings in your config files.

• Switching Drivers: See our Credentials Guide for the one-click sed commands to switch between chan_sip and PJSIP safely.